1. Field of the Invention
The present invention generally relates to a digital circuit multiplication equipment. More specifically, the present invention is directed to a digital circuit multiplication equipment equipped with a tandem pass-through function capable of pass-through-transmitting speech/data signals in a trunk channel connected via a exchange without performing low-bit-rate speech encoding/decoding operations.
2. Description of the Related Art
In long distance telephone communications such as international telephone communications, digital circuit multiplication equipments (DCMEs) have been conducted in order to reduce communication cost. A DCME implies such an apparatus capable of transmitting telephone speech signals and voice band data signals such as facsimile signals and data/modem signals in a higher efficiency by combining a DSI (digital speech interpolation) technique and a low-bit-rate speech encoding technique. This DSI technique corresponds to such a technique capable of transmitting only active portions of telephone communications.
FIG. 16 is a block diagram for schematically representing an entire arrangement of a DCME.
In FIG. 16, there are provided a active judging unit 1, a signal discriminating unit 2, a speech (audio) encoding unit 3, an assignment control unit 4, a message producing unit 5, a multiplexing unit 6, a separating unit 7, a message decrypting unit 8, and a speech decoding unit 9. A left side of this drawing corresponds to a trunk side through which plural channels of telephone speech/voice band data in the 64 Kbit/s PCM (Pulse Code Modulation) format are inputted/outputted. Also, a right side of this drawing corresponds to a bearer side (transmit path side) through which telephone speech/voice band data which have been encoded in low bit rates (will be referred to as “encoded speech signals” hereinafter) are transmitted/received.
For the sake of easy understandings, it is now assumed that while this DCME owns as the trunk side a capacity capable of inputting/outputting 600 channels of 64 Kbit/s-telephone speech/voice band data signals, this DCME owns as the bearer side a line capacity of 2 Mbit/s. Also, the following assumption is made in the below-mentioned description:
As a encoding rate for a low-bit-rate speech encoding operation, a encoding rate of 8 Kbit/s is used so as to transmit a telephone speech signal, whereas another encoding rate of 40 Kbit/s is employed in order to transmit a voice band data signal.
Next, a description will now be made of operations of the DCME equipped with the arrangement shown in FIG. 16.
The 64 Kbit/s PCM-formatted signals for 600 channels, which are entered from the trunk side of this DCME, are inputted into the active judging unit 1, the signal discriminating unit 2, and the speech encoding unit 3, respectively. The active judging unit 1 judges as to whether each of these 600 channels owns an active signal, or a silent signal, and then outputs a judgement result to the assignment control unit 4. The signal discriminating unit 2 judges as to whether an input signal of each trunk channel is equal to a telephone speech (voice) signal, or a data signal such as a facsimile signal, and then outputs this judgement result to the assignment control unit 4.
Based upon both the active/silence judgement result of each of the trunk channels derived from the active judging unit 1 and also the speech/data discrimination result of each of the trunk channels derived from the signal discriminating unit 2, the assignment control unit 4 determines an assignment rate of each trunk channel to the bearer circuit, and then outputs the determined assignment result to both the message producing unit 5 and the multiplexing unit 6.
As this bearer circuit assignment method, while trunk channels having active signals are assigned to bearer circuits while priority rights are given to these trunk channels, the encoding rate of 40 Kbit/s per one channel is assigned to such a trunk channel which is judged as a data channel, and further the encoding rate of 8 Kbit/s per one channel is assigned to such a trunk channel which is judged as a speech channel. The reason why the encoding rate is changed in accordance with a sort of input signal is given as follows: That is, the information amount compression principle of the low-bit-rate speech encoding operation is to reduce redundancy owned by a speech signal, while utilizing redundancy owned by the speech signal. Although the high compression rate can be obtained with respect to the speech signal, the high compression rate cannot be obtained with respect to the voice band data signal such as the facsimile signal.
The encoding rate of each of the trunk channels, which is determined by this assignment control unit 4, is outputted to the speech encoding unit 3, the speech encoding unit 3 is equipped with speech decoding devices for 600 channels. In response to the encoding rate information supplied from the assignment control unit 4, the speech encoding unit 3 encodes the input signals entered from the respective trunk channels by 8 Kbit/s encoding rate when these input signals are the telephone speech signals, and encodes the input signals entered from the respective trunk channels by 40 Kbit/s encoding rate when these input signals are the voice band data signals. Then, the speech encoding unit 3 outputs the encoded signals to the multiplexing unit 6.
It should be noted that the assignment/no-assignment information (namely, information as to whether or not each channel is assigned to a bearer circuit) for the respective channels is also entered from the assignment control unit 4 into the speech encoding unit 3, the reason of which is given as follows:
Normally, both a speech coding device and a speech decoding device are equipped with prediction filters for predicting spectrum information of an input speech signal. The parameter of the prediction filter provided in the speech decoding device which constitutes the transmission side must be made coincident with the parameter of the prediction filter provided in the speech decoding device which constitutes the receiving side. In the case that this prediction filter is such a type of algorithm for updating the past parameter based upon a newly inputted speech signal, in order to make the parameter of the encoding device coincident with the parameter of the decoding device, when the encoding device is connected to the decoding device (namely, when a bearer circuit is newly assigned), both of these encoding/decoding devices are required to be operated from such a condition that these parameters thereof are initialized (reset). This operation is referred to as “a synchronous reset” executed between the speech coding device and the speech decoding device.
As a result, the speech encoding unit 3 may realize this synchronous reset in such a manner that this speech encoding unit 3 initializes the above-explained parameters with respect to the trunk channel operated under such a condition that the bearer has been assigned based upon the assignment/no-assignment information of the bearer circuit, which is changed from another trunk channel condition where the bearer circuit is not assigned. Also, in the speech decoding device, when input operation of the encoded signal from the bearer circuit is commenced, the speech decoding device initializes the parameter thereof.
The message producing unit 5 produces an assignment message to be transmitted to an apparatus on the opposite side on the bearer, based on an assignment result entered from the assignment control unit 4.
In this case, in order to explain the assignment message, FIG. 17 shows a structural example of a frame of a signal (DCME frame) which is outputted by the DCME to the bearer circuit.
In this example, there are 248 channels of bearer channels (BC) and a message channel. The speech encoded data is transmitted via the bearer channels on the bearer circuit. The assignment message is transmitted via the message channel on the bearer circuit. Each of the BC channels owns a capacity of 8 Kbit/s, and therefore, can transmit speech encoded data of 8 Kbit/s corresponding to 248 channels at maximum. Also, the speech encoded data of 40 Kbit/s are transmitted by using 5 channels of the BC channels.
Normally, the length of this DCME frame is selected to be longer than the 8 Kbit/s speech encoding frame length and the 40 Kbit/s speech encoding frame length by an integer multiplication. For example, in the case that the 8 Kbit/s speech encoding frame length is equal to 10 ms and the 40 Kbit/s speech encoding frame length is equal to 2.5 ms, the DCME frame length may be selected to be 10 ms. Subsequently, in this specification, assuming that this length of the DCME frame is equal to 10 ms, the explanation is made (bit number of each BC is equal to 10 ms×8000=0.01 s×8000=80 bits).
Also, 4 sets of messages are transmitted through the message channel. A pair of a trunk channel number (TC number) and a bearer channel number (BC number) may constitute 1 message. For instance, when the number 5 of trunk channel is newly connected to the number 3 of bearer channel, such a message of TC number=5 and BC number=3 is transmitted by employing one message. Normally in the case that TC number=0 indicates “remove”, for example, when the trunk channel connected to BC50 is removed, such a message of TC number=0 and BC number=50 is transmitted.
As previously explained, the assignment message is used to notify such assignment message to the apparatus on the opposite side of the bearer, namely how each of the trunk channels is assigned to the bearer circuit. In order to save the capacity of the message channel, only changing information about the assignment condition is notified as this assignment message. As a consequence, in such a case that there are large change amounts, for example, in the case that a large number of trunk channels are transmitted from silent states to active states at the same time, there is a certain possibility that some channels would wait for assignments to bearer circuits.
Based on the assignment result to the bearer circuit obtained from the assignment control unit 4, the multiplexing unit 6 multiplexes the encoded signals of the respective trunk channels derived from the speech encoding unit 3 to output the multiplexed encoded signal to the bearer circuit. Also, this multiplexing unit 6 multiplexes the assignment messages entered from the message producing unit 5 to output the multiplexed assignment message to the bearer circuit.
Subsequently, a description will now be made of operations of the DCME on the receiving side.
The separating unit 7 inputs thereinto both the encoded signal derived from the bearer circuit and the signal on which the assignment message is multiplexed, and then outputs the assignment message to the message decrypting unit 8 and further outputs the encoded signal to the speech decoding unit 9. It should be noted that when the separating unit 7 separates the encoded signal, the decrypt result of the assignment message is entered from the message decrypting unit 8 so as to separate this encoded signal based upon this decrypt result.
The message decrypting unit 8 inputs the assignment message from the separating unit 7, and then outputs the message decrypt result to the separating unit 7 and at the same time outputs both the assignment/no-assignment information and the encoding rate information of each of the trunk channels to the speech decoding unit 9. The speech decoding unit 9 decodes the encoded signal entered from the separating unit 7 based on the information entered from the message decrypting unit 8 to thereby produce a PCM signal, and then outputs this PCM signal to the respective channels provided on the trunk side.
As previously explained, the DCME encodes the 64 Kbit/s PCM signals supplied from the respective trunk channels by way of the low-bit-rate encoding manner to obtain either the 8 Kbit/s PCM signal or the 40 Kbit/s PCM signal. Furthermore, the DCME transmits such a PCM signal with a top priority, which is judged as a “active” PCM signal. As a consequence, this DCME can transmit the telephone speech signal and the facsimile signal in a low-bit-rate.
On the other hand, the following case will now be considered. As shown in FIG. 18, such DCMEs are arranged at, for example, 3 points, and thus, a network structure is constituted.
Assuming now that the DCMEs are used in the international telephone communication, these points “A”, “B”, and “C” correspond to, for example, communication points in the respective countries such as Japan, USA, and UK. In this network system of FIG. 18, when a telephone communication is established between a telephone set 110 and a telephone set 111, a telephone communication signal produced from the telephone set 110 is low-bit-rate encoded by the DCME 100, and thereafter, this encoded telephone communication signal is decoded by the DCME 101 to produce a PCM signal. This PCM signal is transmitted via a exchange 106 to the DCME 102. In this DCME 102, this PCM signal is again low-bit-rate encoded in a low-bit-rate and then the encoded PCM signal is transmitted to the DCME 103. In this DCME 103, this low-bit-rate encoded signal is decoded to obtain a PCM signal which will then be supplied to the telephone set 111.
As previously explained, when the DCMEs are used in such a network structure as shown in FIG. 18, both the low-bit-rate encoding operation and the decoding operation are repeated two times, resulting in degradation of the telephone communication quality.
To avoid such a problem, such a technique called as “tandem pass-through” has been practically used in the voice ATM communication field and the like.
FIG. 19 is a structural diagram for indicating the voice ATM transmission apparatus 60 equipped with the tandem pass-through function disclosed in Japanese Laid-open Patent Application No. Hei-10-190667.
In FIG. 19, reference numeral 10 shows a cell disassembling unit for disassembling an ATM cell entered from a bearer circuit side to output the disassembled ATM cell, reference numeral 9 represents a speech decoding unit for decoding a encoded signal to output a PCM signal, reference numeral 3 indicates a speech encoding unit for encoding the PCM signal inputted from the trunk side to output a encoded signal, and reference numeral 11 shows a cell assembling unit for assembling the entered encoded signals to output an ATM cell. Also, reference numeral 12 is a pseudo speech signal producing unit for converting either the 8 Kbit/s encoded signal or the 40 Kbit/s encoded signal into such a 64 Kbit/s signal which can be handled by the exchange without executing the decoding operation to output a pseudo speech signal (for example, in case of 8 Kbit/s encoded signal, dummy data of 56 Kbit/s is added to this encoded signal so as to produce pseudo signal of 64 Kbit/s), and reference numeral 13 shows a transmission rate recovering unit for deleting the dummy data of 56 Kbit/s from the pseudo speech signal inputted from the exchange side to convert this pseudo speech signal into the encoded signal having the original encoding rate.
Also, reference numeral 14 indicates a second comfort noise generating unit for generating comfort noise equivalent to background noise during telephone conversation, reference numeral 15 shows a first comfort noise generating unit for generating comfort noise equivalent to background noise during telephone conversation, reference numeral 16 is a first pattern interpolating unit for interpolating a second pattern signal used to cause a voice ATM transmission apparatus as a counter-party apparatus during relay operation to recognize a relay connection, reference numeral 17 is a second pattern interpolating unit for interpolating a first pattern signal used to cause a voice ATM transmission apparatus as a counter-party apparatus during relay operation to recognize such a fact that the own voice ATM transmission apparatus detects the first pattern signal to be brought into relay/switch condition, reference numeral 18 is a first pattern detecting unit for detecting the first pattern signal supplied from the voice ATM transmission apparatus as the counter-party apparatus during relay operation, and also, reference numeral 19 is a second pattern detecting unit for detecting the second pattern signal supplied from the voice ATM transmission apparatus as the counter-party apparatus during relay operation.
Furthermore, reference numeral 20 shows a selector for selectively switching the input signal from the speech encoding unit 3, and the input signal from the transmission rate recovering unit 13, reference numeral 21 indicates a selector for selectively switching the input signal from the first pattern interpolating unit 16 and the input signal from the pseudo speech signal producing unit 12, reference numeral 22 indicates a selector for selectively switching the input signal from the second comfort noise generating unit 14 and the input signal from the pseudo speech signal producing unit 12, reference numeral 23 represents a selector for selectively switching the input signal from the first comfort noise generating unit 15 and the input signal from the selector 20, and reference numeral 24 shows an AND gate circuit for outputting “1” when either one of the input signals derived from the first pattern detecting unit 18 and the second pattern detecting unit 19 becomes “1”, and outputs “0”, if not.
Next, assuming that the voice ATM transmission apparatus 60 equipped with the arrangement shown in FIG. 19 is applied to the positions of the DCME 100, the DCME 101, the DCME 102, and the DCME 103 shown in FIG. 18, operations of this voice ATM transmission apparatus will now be explained.
First, in the case that a telephone communication is established between the telephone set 110 and the telephone set 112 in FIG. 18 (namely, when tandem connection is not made), a description will now be made of operations in the case that the voice ATM transmission apparatus 60 shown in FIG. 18 is installed at the position of the DCME 101.
First of all, in FIG. 19, as the initial condition, the selector 21 selects the input signal from the first pattern interpolating unit 16, the selector 20 selects the input signal from the speech encoding unit 3, the selector 22 selects the input signal from the pseudo speech signal generating unit 12, and the selector 23 selects the input signal from the selector 20, respectively. It should be noted that when the control input signal becomes “0”, these selectors 20, 21, 22, 23 select the input signals of the initial condition side.
In the case that the telephone sets are not tandem-connected by the exchange, both the first pattern detecting unit 18 and the second pattern detecting unit 19 do not detect the first pattern signal and the second pattern signal from the input signal of the trunk side. As a result, the first and second pattern detecting units 18 and 19 output “0” which may indicate that these pattern detecting units 18/19 are not operated under detection states. As a result, the operations of the selectors 20, 21, 22, and 23 are not different from the initial conditions. As a consequence, the speech signal path provided on the transmission side may constitute such a path passing through the speech encoding unit 3, the selector 20, the selector 23, and the cell assembling unit 11. Also, the speech signal path provided on the receiving side may constitute such a path passing through the cell disassembling unit 10, the speech decoding unit 9, the first pattern interpolating unit 16, and the selector 21, so that the normal speech decoding operation and the normal speech encoding operation are carried out.
Now, in the receiving-sided path, the first pattern interpolating unit 16 interpolates the first pattern with respect to the PCM speech signal outputted from the speech decoding unit 9. The PCM signal outputted from the speech decoding unit 9 may become a signal of 64 Kbit/s. That is, this PCM signal corresponds to a signal produced by that the speech signal waveform is sampled every 125 microseconds, and then the amplitude of the sampled waveform is quantized by 8 bits, namely 8/125 microseconds=8/0.000125=64000. In order that the speech quality is not degraded due to this pattern interpolation, the first pattern interpolating unit 16 executes such an operation that only LSB contained in the 8-bit quantized value is bit-stolen every several sampling operations so as to embed a specific pattern into this PCM signal. As a result, even when the first pattern is interpolated, while no adverse influence is given to the original PCM speech signal waveform, the telephone communication can be carried out. The voice ATM transmission apparatus located at the DCME 100 which is connected via the bearer circuit to the DCME 101 as the counter-party apparatus is operated in a similar manner to that of the DCME 101.
Next, a description will now be made of operations of such voice ATM transmission apparatus arranged at the positions of the DCME 101 and the DCME 102 in the case that the telephone sets are relay-connected (tandem-connected) by the exchange, namely, in such a case that a telephone communication is established between the telephone set 110 and the telephone set 111 in FIG. 18.
FIG. 20 is a structural diagram for showing such a case that exchange sides of voice ATM transmission apparatuses are relay-connected. It should be noted that the same reference numerals shown in FIG. 18 and FIG. 19 will be employed as those for denoting the same, or similar circuit components indicated in FIG. 20, and explanations thereof are omitted. In FIG. 20, reference numerals 60B and 60C show two voice ATM transmission apparatuses which constitute a pair, and are connected via a exchange 106 to each other.
When such a connection is made by the exchange, as a first stage, the first pattern detecting unit 18 employed in the voice ATM transmission apparatus 60B detects the first pattern which is interpolated by the first pattern interpolating unit 16 employed in the voice ATM transmission apparatus 60C, and then outputs “1” implying such a signal that the first pattern is detected. Also, the second pattern detecting unit 18 employed in the voice ATM transmission apparatus 60C detects the second pattern which is interpolated by the second pattern interpolating unit 16 employed in the voice ATM transmission apparatus 60B, and then outputs “1” implying such a signal that the second pattern is detected. As a consequence, the conditions of both the voice ATM transmission apparatuses 60B and 60C are changed as follows: The output of the AND gate circuit 24 becomes “1”, the selector 21 selects/outputs the input signal derived from the second pattern interpolating unit 17, the selector 22 selects/outputs the input signal derived from the second comfort noise generating unit 14, and the selector 23 selects/outputs the input signal derived from the first comfort noise generating unit 15. In the voice ATM transmission apparatuses 60B and 60C operated under this condition, the signal path provided on the receiving side may constitute the second comfort noise generating unit 14, the selector 22, the second pattern interpolating unit 17, and the selector 21, whereas the signal path provided on the transmission side may constitute the first comfort noise generating unit 15, the selector 23, and the cell assembling unit 11.
In this case, the second comfort noise generating unit 14 outputs the comfort noise having the 64 Kbit/s PCM format. The second pattern interpolating unit 17 interpolates; the second pattern with respect to the PCM signal outputted from the second comfort noise generating unit 14. In order that this second pattern can be discriminated from the above-explained first pattern, and also does not give a large adverse influence to the outputs of the comfort noise generating unit, the following operation is carried out. For example, only second bit located at the lower bits of the 8-bit quantized value is bit-stolen every several sampling operations with respect to the input PCM signal, and a specific pattern is embedded. As previously explained, the voice ATM transmission apparatuses 60B and 60C may output a silent PCM signal into which the second pattern has been interpolated with respect to the exchange side. Also, the first comfort noise generating unit 15 outputs either 8 Kbit/s encoded silent signal or the comfort noise. As a result, the voice ATM transmission apparatuses 60B and 60C may also output either the silent signal or the comfort noise with regard to the bearer circuit side.
At the next stage, such a silent PCM signal into which the second pattern has been interpolated is entered from the exchange side to the voice ATM transmission apparatuses 60B and 60C. As a result, the second pattern detecting unit 19 detects the second pattern in this silent PCM signal, and thus, outputs “1” indicative of this pattern detection. As a consequence, the selector 20 selects the input signal derived from the transmission rate recovering unit 13 to output the selected signal. Since the first pattern detecting unit 18 does not detect the first pattern, this first pattern detecting unit 18 outputs “0” indicative of no pattern detection. As a result, the condition is changed in such a manner that the selector 23 selects/outputs the input signal derived from the selector 20, and the selector 22 selects/outputs the input signal produced from the pseudo speech signal producing unit.
As to the state of the selector 21, since the output from the AND gate circuit 24 maintains the state of “1” (since second pattern is detected instead of first pattern), this selector 21 maintains such a state that the input signal from the second pattern detecting unit 17 is selected to be outputted. It should be noted that the pseudo speech signal producing unit 12 produces the pseudo speech signal of 64 Kbit/s by adding the dummy bit to the 8 Kbit/s encoded signal entered from the cell disassembling unit. The second pattern is interpolated to a portion of this pseudo speech signal by the second pattern interpolating unit. In other words, the 8 Kbit/s encoded signal is outputted without any problem by assembling the pseudo speech signal in such a way that a portion to be destroyed may constitute a dummy bit. Since this pseudo speech signal is inputted into the transmission rate recovering unit 13, the 8 Kbit/s encoded signal is extracted from this pseudo speech signal to be supplied to the selector 20.
When the apparatus is operated in accordance with the above-explained manner, it can be seen that the pass-through operation can be realized. In other words, in the voice ATM transmission apparatus 60B, the encoded signal whose cells are disassembled by the cell disassembling unit 10 is finally transmitted to the cell assembling unit 11 of the voice ATM transmission apparatus 60C, whereas to the contrary, in the voice ATM transmission apparatus 60C, the encoded signal whose cells are disassembled by the cell disassembling unit 10 is finally transmitted to the cell assembling unit 11 of the voice ATM transmission apparatus 60B.
If the above-described tandem pass-through function is applied also to the DCME shown in FIG. 16, it is expectable that even when the telephone speech signal is transmitted via a plurality of DCME links, this telephone speech signal can be transmitted without degrading the sound quality thereof.
However, when this tandem pass-through technique is applied to a DCME, the below-mentioned problems occur.
For example, the following case will now be considered. That is, in the FIG. 18, the tandem pass-through operation is realized in the case that while a telephone communication is established between the telephone set 110 and the telephone set 111, this communication signal is transmitted through a single trunk channel connected between the DCME 101 and the DCME 102.
In this case, the assignment of the bearer circuit from the DCME 100 to the DCME 101 is changed, depending upon both the active/silence state and the signal discrimination state, which are detected in the DCME 100. For example, when the telephone communication signal produced from the telephone set 100 becomes silent (silent), the connection between the bearer circuit and this trunk channel along the direction defined from the DCME 100 to the DCME 101 may be removed. This information on bearer assignment may be notified by embedding the active/silence information into the pseudo speech signal which is transmitted from the DCME 101 to the DCME 102. Then, the DCME 102 may determine the assignments of this trunk channel to the bearer circuit based upon the active/silence information embedded in this pseudo speech signal.
However, when the bearer circuit is assigned, such a channel operated under silent state is searched from the trunk channels under connection with the bearer circuit, and thereafter, the bearer circuit assignment must be rearranged. Assuming now that all of the trunk channels connected to the bearer circuits are operated under active states, the bearer circuit assignments of a trunk channel witch newly become under active state may be waited. When the bearer circuit assignment is brought into the waiting state, a portion of the “active” speech signal is dropped, and therefore, a so-called “freeze out” phenomenon will occur. In general, if the time rate of the “freeze out” phenomenon occurs with respect to the entire active time, namely freeze out fraction ratio is smaller than, or equal to 0.5 percents, substantially no detection can be made of degradations in telephone communications.
However, when such a freeze out phenomenon occurs in the trunk channel operated under tandem pass-through connection, not only a portion of the speech signal is dropped out, but also another problem may be newly produced. That is, the speech coding device and the speech decoding device cannot be reset in the synchronous resetting mode. It should also be noted that such a freeze out phenomenon may be produced by limiting a total number of messages.
As previously explained, when the synchronous resetting operation cannot be realized between the speech coding device and the speech decoding device, the internal parameters of both devices are not made coincident with each other. As a result, degradations of the telephone communication quality may be considerably induced.
Furthermore, a similar inconvenient case may occur in such a case that the speech encoding rate is changed, due to the change in the speech/data discrimination state, and thus, the bearer circuit assignment rate is changed.
For instance, considering now such a condition that the bearer circuit assignment rate from the DCME 100 to the DCME 101 is changed from 8 Kbit/s to 40 Kbit/s, the tandem pass-through function may be expectedly realized when the following operation is basically carried out. That is, similar to the active/silence information, while the encoding rate information is embedded in the pseudo speech signal, the bearer circuit assignment is determined based on this information in the DCME 102.
However, in such a case that a change in the bearer circuit assignment rate is brought into a waiting state in the DCME 102 due to a lack of total bearer circuit capacity to be assigned and also a limitation in total message number, such a problem may occur in which in this trunk channel, there is no signal to be outputted to the bearer circuit (namely, bearer circuit assignment is selected to be 8 Kbit/s whereas encoded data extracted from pseudo speech signal is 40 Kbit/s). As a result, while such a problem is continued, a certain invalid signal should be necessarily outputted to the DCME 103. As a consequence, there is a certain possibility that extraordinary sounds may be reproduced from the speech decoding device in the DCME 103.
Also, when the bearer circuit assignment rate is changed into 40 Kbit/s after a certain time period has passed, since the encoded data are lost by such a data quantity corresponding to the waiting time period for the assignment change, the internal parameter of the speech decoding device employed in the DCME 100 is not made coincident with the internal parameter of the speech decoding device employed in the DCME 103, resulting in degradation in the telephone communication quality.
As described above, in the conventional technique, when the tandem pass-through function is attempted to be installed in DCME, the bearer circuit assignment change is delayed, with the result that a speech coding device and a speech decoding device can not be reset in a synchronous resetting mode in DCME provided on both ends of the transmission path, which causes disagreement of the internal parameter thereof, and arises a problem of degradation in the telephone communication quality.